JP2007329631A - Acoustic correction device - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide an acoustic correction device capable of providing sufficient sound effect without making the filter length of an FIR filter long. <P>SOLUTION: The acoustic correction device 3 comprises acoustic correction means 10a to 10d of generating a sound signal outputting only an initial reflected sound from an output signal of an output sound based upon a transfer function from an output position of the output sound to the right and left ears of a listener, reverberation processing means 11a and 11b of generating sound signals outputting only reverberation sounds from the output signal of the output sound, and adding means 13a and 13b of generating outputting sound signals outputting the initial reflected sound and reverberation sound by adding the sound signals output by the reverberation processing means 11a and 11b together. <P>COPYRIGHT: (C)2008,JPO&INPIT

Description

  The present invention relates to an acoustic correction apparatus including an acoustic correction unit that performs an acoustic correction process on an acoustic signal of an output sound based on a transfer function from an output position of the output sound to left and right ears of a listener.

  The acoustic environment in the passenger compartment is characterized by the intrusion of vehicle-specific noise such as road noise, and because the passenger compartment is formed by a unique narrow closed space, the acoustic environment differs from that of the normal acoustic space. It is characterized by having characteristics. For this reason, in order to construct a suitable acoustic environment in the vehicle interior, it is necessary to appropriately correct the output sound output from the in-vehicle device such as a car stereo.

  As a method of correcting the output sound of the in-vehicle device, generally, a digital filter (FIR filter (Finite Impulse Response Filter) is used to perform sound correction processing of a sound source such as a CD (Compact Disc) or a DVD (Digital Versatile Disk). The method is used.

  FIG. 5A is a diagram illustrating a state in which the listener L listens to output sound output from the speakers 50 a and 50 b that are installed symmetrically with respect to the front direction of the listener L. Since the output sound output from the left speaker 50a reaches the listener L as well as the right ear as well as the left ear, the listener L must listen to the output sound of the left speaker 50a from both the left and right ears. It becomes. Similarly, since the output sound output from the right speaker 50b reaches the listener L as well as the right ear as well as the left ear, the listener L outputs the right speaker 50b from both the left and right ears. You will hear the sound.

  The sound pressure level of the sound emitted from the left and right speakers 50a and 50b varies depending on the frequency before reaching the eardrum via space, head to ear. The transfer characteristic of sound from the sound source to the ear is called a transfer function (head-related transfer function). The listener L can specify the position of the sound source because the listener L knows its own head-related transfer function and its angular dependency. Specifically, the listener L can determine the sound pressure level of the perceived sound. The angle of the sound source is determined from the frequency characteristics and the transfer function of the head. Therefore, the acoustic environment of the sound that the listener L listens to can be changed by acoustically correcting the audio signals output from the left and right in consideration of this transfer function (see, for example, Patent Document 1).

FIG. 5B is a block diagram illustrating a schematic configuration of a conventionally used acoustic correction apparatus. The acoustic correction device 51 includes four digital filters 52a to 52d and two adders 53a and 53b. FIR filters are used for the digital filters 52a to 52d. The FIR filter is a filter that converges an output signal when impulse input is performed to 0 in a finite time, and is represented by Expression (1).

.... Formula (1)
Here, y [n] is an output signal, x [n] is an input signal, and h [i] is a transfer function.

  The expression (1) indicates that the output signal y [n] at time n is the current and past input signal values x [n], x [n−1], x [n−2],. n−N] and x [n− (N−1)] ”, and N shown in Expression (1) is called a filter length (filter coefficient length).

  When applying an acoustic effect to an audio signal, measure the impulse response in the space where the acoustic effect is to be reproduced, such as a concert hall, and obtain the transfer function of the acoustic space by Fourier transforming the measured impulse response. Build an FIR filter.

  FIG. 6 is a block diagram showing a schematic configuration of the FIR filter. The FIR filter 55 includes a delay circuit 56 connected continuously in multiple stages, an input signal, and an output signal delayed by each delay circuit 56. A multiplier 57 that multiplies each by a different coefficient, and an adder 58 that sequentially adds the multiplied signals. The coefficient multiplied by the multiplier is set according to the transfer function.

  The multiplier coefficients (filter coefficients) of the digital filters 52a to 52d shown in FIG. 5B are set based on the four transfer functions described with reference to FIG. Specifically, the digital filter 52a uses a transfer function HLL (first transfer function) when sound is transmitted from the left speaker 50a shown in FIG. 5A to the left ear of the listener L, and the digital filter 52b. Is used as a transfer function HLR (second transfer function) when sound is transmitted from the left speaker 50a to the right ear of the listener L, and the digital filter 52c has a sound from the right speaker 50b to the left ear of the listener L. Is used, and the transfer function HRR (fourth transfer function) when sound is transmitted from the right speaker 50b to the right ear of the listener L is used for the digital filter 52d. It is done.

  As shown in FIG. 5B, the audio signal output from the left speaker 50a is subjected to acoustic correction processing when it is transmitted from the left speaker 50a to the left ear by the digital filter 52a, and the digital filter 52b. Thus, an acoustic correction process is performed when the signal is transmitted from the left speaker to the right ear. The audio signal output from the right speaker 50b is subjected to acoustic correction processing when it is transmitted from the right speaker 50b to the left ear by the digital filter 52c, and from the right speaker to the right ear by the digital filter 52d. An acoustic correction process for transmission is performed.

  The acoustic signals corrected for the left ear by the digital filter 52a and the digital filter 52c are added by the adder 53a and output as an output signal L for the left ear. The acoustic signals corrected for the right ear by the digital filter 52b and the digital filter 52d are added by the adder 53b and output as an output signal R for the right ear.

As described above, the acoustic environment of the concert hall or the like can be constructed by performing the acoustic correction of the output sound by the digital filters 52a to 52d using the FIR filter 55.
JP-A-7-336798 (page 5-7, FIG. 1)

  When acoustic correction is performed using the above-described FIR filter, it is necessary to sufficiently increase the filter length of the FIR filter in order to obtain a sufficient acoustic effect.

  FIG. 7A shows an impulse response when the filter length is 4096. FIG. In the impulse response with a filter length of 4096, as shown in FIG. 7A, the initial reflected sound portion A and the reverberant sound portion B appear in the waveform, and the listener L listens to the initial reflected sound portion A. By judging the output direction of the output sound by listening to the reverberation sound part B, the spatial extent of the output sound can be felt.

  When such acoustic correction processing is performed by a hardware circuit, if the filter length is increased, the number of delay circuits 56, multipliers 57, and adders 58 increases as shown in FIG. There is a problem that the hardware circuit becomes expensive.

  In addition, even when sound correction is performed by software processing using a DSP (Digital Signal Processor) or the like, there is a problem in that the amount of memory required for calculation increases and calculation time increases. .

  On the other hand, when the filter length is simply shortened, there is a problem that a sufficient acoustic effect cannot be obtained. FIG. 7B shows an impulse response when the filter length is 128. In an impulse response with a filter length of 128, as shown in FIG. 7B, the initial reflected sound portion A appears in the waveform, but the reverberation sound portion B is not shown at all. For this reason, the listener L can determine the output direction of the output sound by listening to the initial reflected sound portion A, but cannot listen to the reverberation sound portion B, so that the spatial extent of the output sound is increased. I couldn't feel it, and I couldn't get enough acoustic environment.

  The present invention has been made in view of the above-described problems, and an object of the present invention is to provide an acoustic correction apparatus capable of producing a sufficient acoustic effect without increasing the filter length of the FIR filter.

  In order to solve the above problems, an acoustic correction device according to the present invention is configured to generate only an initial reflected sound from an output signal of the output sound based on a transfer function from the output position of the output sound to the left and right ears of the listener. An acoustic correction means for generating an acoustic signal to be output; a reverberation processing means for generating an acoustic signal for outputting only a reverberation sound from the output signal of the output sound; an acoustic signal generated by the acoustic correction means and the reverberation processing means; And an adding means for generating an acoustic signal that adds the acoustic signal generated by the step (1) to output the initial reflected sound and the reverberant sound.

  The acoustic correction unit may be configured by an FIR filter, and may generate an acoustic signal that outputs only the initial reflected sound by shortening the filter length of the FIR filter.

  Furthermore, the acoustic correction means includes a first FIR filter in which a filter coefficient is set based on a first transfer function from the output signal for the L channel to the left ear of the listener, and the output for the L channel. A second FIR filter in which a filter coefficient is set based on a second transfer function from the signal to the listener's right ear, and a third transfer function from the R channel output signal to the listener's left ear And a fourth FIR filter in which the filter coefficient is set based on the fourth transfer function from the output signal for the R channel to the right ear of the listener. A reverberation processing means for performing reverberation processing on the output signal for the L channel, and a reverberation processing means for the output signal for the R channel. Second reverb processing means for performing sound processing, wherein the adding means outputs an output signal generated via the first FIR filter and an output signal generated via the third FIR filter. And an output signal generated via the first reverberation processing means, and a first addition means for generating an output signal for L channel and the second FIR filter. The output signal generated through the fourth FIR filter and the output signal generated through the second reverberation processing unit are added to generate an output signal for the R channel. And a second addition means that sets the filter length of the first to fourth FIR filters to a short value.

  The acoustic correction device according to the present invention is generated by the acoustic correction unit that generates the acoustic signal that outputs only the initial reflected sound, the reverb processing unit that generates the acoustic signal that outputs only the reverberation sound, and the acoustic correction unit. And an adding means for generating an acoustic signal that outputs an initial reflected sound and a reverberation sound by adding the acoustic signal generated by the reverberation processing means and the acoustic signal generated by the reverberation processing means. Among them, the output direction of the output sound can be reproduced by the initial reflected sound part, and the spatial spread of the output sound can be reproduced by the reverberant sound part.

  Further, the filter length of the FIR filter is shortened to generate an acoustic signal that outputs only the initial reflected sound by the FIR filter, and further, the reverb processing means generates an acoustic signal that outputs only the reverberant sound, thereby reducing the filter length. The effect equivalent to the acoustic correction effect that can be obtained by the FIR filter by increasing the length can be realized by using the FIR filter having a short filter length. As described above, since the filter length of the FIR filter can be made shorter than before, an increase in circuit scale accompanying the acoustic correction processing of the FIR filter can be suppressed, and the memory capacity necessary for the calculation can be greatly reduced. be able to.

  Furthermore, by using reverberation processing means, the circuit configuration is simplified compared to the case where acoustic correction processing is performed using an FIR filter having a long filter length, and the same acoustic effect is achieved while reducing the memory required for arithmetic processing. Therefore, it is possible to achieve a sufficient acoustic effect with a simple configuration.

  Hereinafter, an acoustic correction system including an acoustic correction apparatus according to the present invention will be described in detail with reference to the drawings.

  FIG. 1 is a block diagram showing a schematic configuration of an acoustic correction system. The sound correction system 1 includes an audio reproduction unit 2, a sound correction unit (sound correction device) 3, speakers 4 a and 4 b, a memory 5, an operation unit 6, and a control unit 7.

  The audio playback unit 2 has a role of reading sound source data recorded on a recording medium such as a CD or DVD as an input signal and outputting the input signal to the sound correction unit 3. The sound source data read by the audio playback unit 2 is not necessarily limited to that recorded on a recording medium such as a CD or DVD, and may be sound source data recorded on a hard disk drive or the like. Further, instead of the audio playback unit 2, the sound correction system 1 may be provided with an external input terminal (not shown) and the like, and the sound source data played back by another audio device may be read through this external input terminal. Good. In the present embodiment, it is assumed that the audio reproduction unit 2 acquires acoustic data for L channel and acoustic data for R channel, and outputs them as an input signal L and an input signal R.

  The speakers 4a and 4b are devices that output the output signal L and the output signal R corrected by the acoustic correction unit 3 as output sounds. The speaker 4a is an L channel speaker and the speaker 4b is an R channel speaker. Is shown. The speakers 4a and 4b are configured by a unit including a cone, a magnet, a voice coil, and the like, like a general speaker.

  The operation unit 6 is an operation means for the listener to select a desired sound correction mode, for example, a sound environment such as a concert hall mode or a stadium mode. In accordance with the acoustic correction mode selected by the listener using the operation unit 6, the control unit 7 performs acoustic correction setting of the acoustic correction unit 3. The memory 5 stores information (for example, a transfer function) related to the acoustic correction mode used when the acoustic correction unit 3 performs the acoustic correction process, and when the acoustic correction setting is performed in the control unit 7. The information regarding the acoustic correction mode stored in the memory 5 is used.

  The control unit 7 has a function of determining a multiplier coefficient (filter coefficient) of an FIR filter, which will be described later, according to the acoustic correction mode selected by the listener, and sending it to the acoustic correction unit 3 as a processing parameter for the acoustic correction process. ing. Specifically, the control unit 7 acquires the acoustic correction mode information selected by the listener from the operation unit 6, and reads related information from the memory 5 according to the acquired acoustic correction mode. Then, based on the read information, the coefficient value of the FIR filter is determined and the processing parameter is transmitted.

  The sound correction unit 3 has a role of outputting to the speakers 4 a and 4 b after performing sound correction processing on the input signal L and the input signal R acquired from the audio reproduction unit 2. FIG. 2 is a block diagram showing a schematic configuration of the acoustic correction unit 3.

  The acoustic correction unit 3 includes four digital filters (acoustic correction means) 10a to 10d, two reverberation processors (reverb processing means: first reverb processing means, second reverb processing means :) 11a, 11b, and 2 There are two multipliers 12a, 12b and four adders (adding means: first adding means, second adding means) 13a to 13d.

  FIR filters (first to fourth FIR filters) are used for the digital filters 10 a to 10 d, and the digital filters 10 a to 10 d are received from the audio reproducing unit 2 based on the processing parameters received from the control unit 7. It has a role of performing acoustic correction processing on the received input signal L and input signal R. The FIR filter is configured by a delay device, a multiplier, and an adder in the same manner as the FIR filter already described with reference to FIG. The coefficient in the multiplier is set by a processing parameter received from the control unit 7.

  Specifically, a coefficient of a processing parameter set based on a transfer function when sound is transmitted from the left speaker to the left ear of the listener is used for the digital filter 10a, and the digital filter 10b is used for the listener from the left speaker. The coefficient of the processing parameter set based on the transfer function when sound is transmitted to the right ear is used, and the digital filter 10c is based on the transfer function when sound is transmitted from the right speaker to the left ear of the listener. The processing parameter coefficients set based on the transfer function when sound is transmitted from the right speaker to the right ear of the listener is used for the digital filter 10d. However, the filter length of the FIR filter according to the present embodiment is set to a relatively short value, for example, 128.

  The reverb processors 11 a and 11 b have a role of performing reverberation processing (Reverbration processing) on the input signal L and the input signal R received from the audio reproduction unit 2. In the reserve processors 11a and 11b, as shown in FIG. 3A, one unit is formed by one delay circuit 15, two multipliers 16a and 16b, and two adders 17a and 17b. As necessary, as shown in FIG. 3 (b), a plurality of adders 17a and 17b are combined to be connected in series. In the present embodiment, the reverb processing machines 11a and 11b are connected to three units. The multiplier 12a and the multiplier 12a have a role of adjusting the amplitudes of the output signal L and the reverberation signal R subjected to the reverberation processing by the reverb processors 11a and 11b.

  Next, the sound correction unit 3 performs sound correction processing on the input signal L / R input signal R input from the audio reproduction unit 2, and then outputs the output signal L / output signal R to the speakers 4a, 4b. A processing procedure will be described.

  First, as shown in FIG. 2, the input signal L is input to the digital filter 10a, the digital filter 10b, and the reverb processor 11a, and the input signal R is input to the digital filter 10c, the digital filter 10d, and the reverb processor 11b. Entered.

  The input signal L input to the digital filter 10a is subjected to acoustic correction processing when it is transmitted from the left speaker to the left ear by the digital filter 10a, and the input signal L input to the digital filter 10b is converted to the digital filter 10b. Thus, an acoustic correction process is performed when the signal is transmitted from the left speaker to the right ear.

  The input signal R input to the digital filter 10c is subjected to acoustic correction processing when it is transmitted from the right speaker to the left ear by the digital filter 10c, and the input signal R input to the digital filter 10d is digital An acoustic correction process is performed when the signal is transmitted from the right speaker to the right ear by the filter 10d.

  Then, the acoustic signals corrected for the left ear by the digital filter 10a and the digital filter 10c are added in the adder 13a to become an output signal La for the left ear. Further, the acoustic signals corrected for the right ear by the digital filter 10b and the digital filter 10d are added together in the adder 13b to become an output signal Ra for the right ear.

  Since the acoustic correction processing performed by each of the digital filters 10a to 10d is performed based on processing parameters determined based on the acoustic correction mode selected by the operation unit 6, acoustic correction is performed by the digital filters 10a to 10d. By outputting the output signal from the speakers 4a and 4b, an acoustic environment such as a concert hall can be constructed.

  However, since the filter length in the digital filters 10a to 10d is set to 128 as described above, the impulse response of the output signal to which the FIR filter is applied in the digital filters 10a to 10d is as shown in FIG. Only the initial reflected sound portion A is shown.

  On the other hand, the input signal L input to the reverb processor 11a and the input signal R input to the reverb processor 11b are subjected to reverberation processing by the reverb processors 11a and 11b.

  Specifically, the input signal L input to the reverb processor 11 a is divided into a signal delayed by the delay circuit 15 and a signal multiplied by a predetermined coefficient by the multiplier 16 a without going through the delay circuit 15. The signal delayed by a predetermined time by the delay circuit 15 is fed back via the multiplier 16b and added to the input signal by the adder 17a and the signal multiplied by the adder 17b by the multiplier 16a. Are divided into signals to be output signals.

  In this way, feedback and the like are performed after being delayed by the delay circuit 15 and the signals are superimposed, so that the reverb processor 11a has an effect that a plurality of reflected sounds caused by the delay overlap and are combined, and reverberation. There will be an effect. The output signal subjected to the reverberation processing by the reverb processor 11a is adjusted in amplitude by the multiplier 12a to become an output signal Lb. Thereafter, the output signal Lb is added by the adder 13c to the output signal La subjected to the acoustic correction processing by the digital filters 10a and 10c, and is output to the speaker 4a as the output signal L for the L channel.

  Similarly, the input signal R input to the reverberation processor 11b is subjected to reverberation processing and then added to the output signal Ra subjected to acoustic correction processing by the digital filters 10b and 10d by the adder 13d. The output signal R for the channel is output to the speaker 4b.

  FIG. 4B shows an impulse response of the output signal Lb subjected to reverberation processing by giving a predetermined delay number and coefficient as parameters to the delay circuit 15 and the multipliers 16a and 16b of the reverb processor 11a. Is shown. As shown in FIG. 4B, the output signal Lb subjected to the reverberation processing does not represent the initial reflected sound portion A but represents the reverberation sound portion B. Therefore, the adder 13c adds the output signal La that has been subjected to the acoustic correction processing by the digital filters 10a and 10c and the output signal Lb that has been subjected to the reverberation processing by the reverb processing unit 11a, so that FIG. As shown in c), an impulse response including an initial reflected sound portion A and a reverberation sound portion B can be obtained.

  In particular, by appropriately setting the coefficient of the FIR filter, the delay number and coefficient of the reverberation unit 11a, and the amplitude of the output signal from the multiplier 12a, even if the filter length of the FIR filter is about 128, the filter length can be several thousand. An impulse response similar to that obtained when the above FIR filter is used alone can be obtained.

  In this way, by performing the acoustic correction process using the FIR filters (digital filters 10a to 10d) having a short filter length and the reverb processors 11a and 11b, the initial reflected sound portion A subjected to the acoustic correction process by the FIR filter is used. The output direction of the output sound can be reproduced, and the spatial expansion of the output sound can be reproduced by the reverberation sound part B subjected to the reverberation sound processing by the reverb processors 11a and 11b.

  Furthermore, when the acoustic correction effect obtained by increasing the filter length in the FIR filter is realized by using the FIR filters (digital filters 10a to 10d) having a short filter length and the reverb processors 11a and 11b, the FIR filter Since the same effect can be obtained with a configuration (3 unit length in this embodiment) in which the connecting unit length of the reverb processing units 11a and 11b is significantly shortened compared to the filter length of the filter, an increase in circuit scale is suppressed. The memory capacity required for the calculation can be greatly reduced.

  Since the memory capacity required for the acoustic correction process can be reduced, a plurality of filter coefficients of the FIR filter and a plurality of reverb processors can be used instead of the memory 5 shown in FIG. 1 if necessary. A database in which parameters (number of delays and coefficients) are combined and recorded as a parameter set may be provided. The listener can freely select a parameter set via the operation unit 6 and switches a filter coefficient or the like from the database based on the parameter set, or based on the parameter set at an arbitrary timing according to an input signal to be input. It is also possible to switch the filter coefficient or the like.

  As described above, by using the acoustic correction apparatus according to the present invention, it is possible to achieve a sufficient acoustic correction effect without increasing the filter length in the FIR filter, and to reduce the memory usage capacity and the calculation time. It becomes possible to shorten.

  In addition, although the acoustic characteristic correction apparatus which concerns on this invention was demonstrated in detail using drawing, the acoustic characteristic correction apparatus which concerns on this invention is not limited to embodiment mentioned above. It will be apparent to those skilled in the art that various changes and modifications can be made within the scope of the claims, and these are naturally within the technical scope of the present invention. Is done.

It is the block diagram which showed schematic structure of the acoustic correction system which concerns on this embodiment. It is the block diagram which showed schematic structure of the acoustic correction part which concerns on this embodiment. It is the block diagram which showed schematic structure of the reverberation processor which concerns on this embodiment. (A) is a graph which shows the impulse response of the output signal La concerning this embodiment, (b) is a graph which shows the impulse response of the output signal Lb concerning this embodiment, (c) is this graph It is a graph which shows the impulse response of the output signal L which concerns on embodiment. (A) is the figure which showed the structure by which the speaker etc. were installed symmetrically with respect to the front direction of a listener, (b) is the block diagram which showed schematic structure of the conventional acoustic correction apparatus. It is the block diagram which showed schematic structure of the general FIR filter. (A) is the graph which showed the impulse response of the output signal by which the acoustic correction process was made by the FIR filter with a long filter length, (a) is the impulse of the output signal by which the acoustic correction process was made by the FIR filter with a short filter length It is the graph which showed the response.

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 ... Sound correction system 2 ... Audio reproduction part 3 ... Sound correction part (acoustic correction apparatus)
4a, 4b, 50a, 50b ... speaker 5 ... memory (database)
6 ... operation part 7 ... control part 10a, 10b, 10c, 10d ... digital filter (acoustic correction means, FIR filter)
11a, 11b ... Reverb processor (reverb processing means)
12a, 12b ... (acoustic correction unit) multipliers 13a, 13b, 13c, 13d ... (acoustic correction unit) adders (adding means)
15 ... delay circuit 16a, 16b (for reverb processor) ... multiplier 17a, 17b (for reverb processor) ... adder 51 (for reverb processor) ... (conventional) acoustic correction device 52a, 52b, 52c, 52d ... digital filters 53a, 53b (of a conventional acoustic correction apparatus) ... adders 55 (of a conventional acoustic correction apparatus) ... FIR filters 56 ... delay circuits 57 (of FIR filters) ... multipliers 58 (of FIR filters) ( Adder L of FIR filter (listener)

Claims (3)

Based on a transfer function from the output position of the output sound to the left and right ears of the listener, acoustic correction means for generating an acoustic signal that outputs only the initial reflected sound from the output signal of the output sound;
Reverberation processing means for generating an acoustic signal for outputting only reverberation sound from the output signal of the output sound;
Addition means for adding the acoustic signal generated by the acoustic correction means and the acoustic signal generated by the reverberation processing means to generate an acoustic signal that outputs the initial reflected sound and the reverberation sound. A characteristic acoustic correction apparatus.   The acoustic correction device according to claim 1, wherein the acoustic correction unit includes an FIR filter, and generates an acoustic signal that outputs only the initial reflected sound by shortening a filter length of the FIR filter. . The acoustic correction means includes a first FIR filter in which a filter coefficient is set based on a first transfer function from the L channel output signal to the left ear of the listener, and the L channel output signal. Based on the second FIR filter in which the coefficient of the filter is set based on the second transfer function up to the right ear of the listener and the third transfer function from the output signal for the R channel to the left ear of the listener A third FIR filter in which the filter coefficient is set, and a fourth FIR filter in which the filter coefficient is set based on a fourth transfer function from the output signal for the R channel to the right ear of the listener And
The reverberation processing means includes first reverberation processing means for performing reverberation processing on the L channel output signal, and second reverberation processing means for performing reverberation processing on the R channel output signal. And
The addition means is generated via the output signal generated via the first FIR filter, the output signal generated via the third FIR filter, and the first reverberation processing means. First output means for adding an output signal to generate an output signal for the L channel, an output signal generated via the second FIR filter, and generated via the fourth FIR filter And a second addition means for adding the output signal generated through the second reverberation processing means to generate an output signal for the R channel,
The acoustic correction apparatus according to claim 1, wherein a filter length of the first to fourth FIR filters is set to a short value.